mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
fde2b24281
Potential deadlock fixed by acquiring lock before calling encoder.
This is a reland of a135557b3c
Original change's description:
> Call OnReceivedOverhead after audio network adaptor is created.
>
> This prevents ending up in a state where audio network adaptor never
> receives the current packet overhead and therefore doesn't work.
>
> Bug: chromium:1086942
> Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31951}
Bug: chromium:1086942
Change-Id: I514e523c6607cee0099b87919f0f77ebec966ddd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181888
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31971}
116 lines
4.2 KiB
C++
116 lines
4.2 KiB
C++
/*
|
|
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef TEST_MOCK_AUDIO_ENCODER_H_
|
|
#define TEST_MOCK_AUDIO_ENCODER_H_
|
|
|
|
#include <string>
|
|
|
|
#include "api/array_view.h"
|
|
#include "api/audio_codecs/audio_encoder.h"
|
|
#include "test/gmock.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class MockAudioEncoder : public AudioEncoder {
|
|
public:
|
|
MockAudioEncoder();
|
|
~MockAudioEncoder();
|
|
MOCK_METHOD(int, SampleRateHz, (), (const, override));
|
|
MOCK_METHOD(size_t, NumChannels, (), (const, override));
|
|
MOCK_METHOD(int, RtpTimestampRateHz, (), (const, override));
|
|
MOCK_METHOD(size_t, Num10MsFramesInNextPacket, (), (const, override));
|
|
MOCK_METHOD(size_t, Max10MsFramesInAPacket, (), (const, override));
|
|
MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
|
|
MOCK_METHOD((absl::optional<std::pair<TimeDelta, TimeDelta>>),
|
|
GetFrameLengthRange,
|
|
(),
|
|
(const, override));
|
|
|
|
MOCK_METHOD(void, Reset, (), (override));
|
|
MOCK_METHOD(bool, SetFec, (bool enable), (override));
|
|
MOCK_METHOD(bool, SetDtx, (bool enable), (override));
|
|
MOCK_METHOD(bool, SetApplication, (Application application), (override));
|
|
MOCK_METHOD(void, SetMaxPlaybackRate, (int frequency_hz), (override));
|
|
MOCK_METHOD(void,
|
|
OnReceivedUplinkBandwidth,
|
|
(int target_audio_bitrate_bps,
|
|
absl::optional<int64_t> probing_interval_ms),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
OnReceivedUplinkPacketLossFraction,
|
|
(float uplink_packet_loss_fraction),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
OnReceivedOverhead,
|
|
(size_t overhead_bytes_per_packet),
|
|
(override));
|
|
|
|
MOCK_METHOD(bool,
|
|
EnableAudioNetworkAdaptor,
|
|
(const std::string& config_string, RtcEventLog*),
|
|
(override));
|
|
|
|
// Note, we explicitly chose not to create a mock for the Encode method.
|
|
MOCK_METHOD(EncodedInfo,
|
|
EncodeImpl,
|
|
(uint32_t timestamp,
|
|
rtc::ArrayView<const int16_t> audio,
|
|
rtc::Buffer*),
|
|
(override));
|
|
|
|
class FakeEncoding {
|
|
public:
|
|
// Creates a functor that will return |info| and adjust the rtc::Buffer
|
|
// given as input to it, so it is info.encoded_bytes larger.
|
|
explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
|
|
|
|
// Shorthand version of the constructor above, for when only setting
|
|
// encoded_bytes in the EncodedInfo object matters.
|
|
explicit FakeEncoding(size_t encoded_bytes);
|
|
|
|
AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
|
|
rtc::ArrayView<const int16_t> audio,
|
|
rtc::Buffer* encoded);
|
|
|
|
private:
|
|
AudioEncoder::EncodedInfo info_;
|
|
};
|
|
|
|
class CopyEncoding {
|
|
public:
|
|
~CopyEncoding();
|
|
|
|
// Creates a functor that will return |info| and append the data in the
|
|
// payload to the buffer given as input to it. Up to info.encoded_bytes are
|
|
// appended - make sure the payload is big enough! Since it uses an
|
|
// ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
|
|
// of scope!
|
|
CopyEncoding(AudioEncoder::EncodedInfo info,
|
|
rtc::ArrayView<const uint8_t> payload);
|
|
|
|
// Shorthand version of the constructor above, for when you wish to append
|
|
// the whole payload and do not care about any EncodedInfo attribute other
|
|
// than encoded_bytes.
|
|
explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
|
|
|
|
AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
|
|
rtc::ArrayView<const int16_t> audio,
|
|
rtc::Buffer* encoded);
|
|
|
|
private:
|
|
AudioEncoder::EncodedInfo info_;
|
|
rtc::ArrayView<const uint8_t> payload_;
|
|
};
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // TEST_MOCK_AUDIO_ENCODER_H_
|